Stephan M. Bernsee wrote:

> That's the plan.

Yowser!  :-)

Oh, and while I'm at it, how about leaving the sample values 
and the first derivative unchanged at each end (for 
catenating separately scaled regions.)

Ah, I'm sure I'll come up with more.  I suppose the best 
place for this would be your site.  How's about adding a 
click box for "Wish List".


Bob
-- 

"Things should be described as simply as possible, but no 
simpler."

                                              A. Einstein

Yes, you are right.  That was a mistake on my part due to being tired .

I've found a set of classes that does what i need.  Thanks for all the help
and brainstorming.

David

"Google User" <someonehr@yahoo.com> wrote in message
news:44fa6d0f.0404220208.4da3f759@posting.google.com...
> "David Reid" <dreid_nospam@remove_no_spam_mechtronix.ca> wrote in message
news:<mDAhc.57875$Gp4.1315198@news20.bellglobal.com>...
> > Well, i think that's what DirectSound allows you to do when you change
the
> > buffer frequency.  If the sample rate of the sound is 44100 and you
multiply
> > by 1.08 approx, the buffer will now contain data for 48kHz playback,
which
> > when played back at 44.1kHz would yield a "faster" sound.  This method
was
> > acceptable for the company so that's what im trying to achieve but using
a
> > different sound API (ASIO).  So i have to do the sample rate conversions
> > myself.
>
> Maybe, I am misunderstanding something here. If you have original data
> at 44.1K, then you upsample it at 48K and then if you play it back
> again at 44.1K, it would yield a slower sound, not faster. Maybe, what
> you should do is play the data at 44.1K at 48K to shift the pitch
> higher and for the sound to go faster?
>
> >
> > I've started looking at the osalp on sourceforge, it has a class that
does
> > sample rate conversions.  i think that will be good enough for what i
want
> > to do, but we'll see how it goes.
> >
> > Thanks for all your help.

"David Reid" <dreid_nospam@remove_no_spam_mechtronix.ca> wrote in message news:<mDAhc.57875$Gp4.1315198@news20.bellglobal.com>...
> Well, i think that's what DirectSound allows you to do when you change the
> buffer frequency.  If the sample rate of the sound is 44100 and you multiply
> by 1.08 approx, the buffer will now contain data for 48kHz playback, which
> when played back at 44.1kHz would yield a "faster" sound.  This method was
> acceptable for the company so that's what im trying to achieve but using a
> different sound API (ASIO).  So i have to do the sample rate conversions
> myself.

Maybe, I am misunderstanding something here. If you have original data
at 44.1K, then you upsample it at 48K and then if you play it back
again at 44.1K, it would yield a slower sound, not faster. Maybe, what
you should do is play the data at 44.1K at 48K to shift the pitch
higher and for the sound to go faster?

> 
> I've started looking at the osalp on sourceforge, it has a class that does
> sample rate conversions.  i think that will be good enough for what i want
> to do, but we'll see how it goes.
> 
> Thanks for all your help.

That's the plan.

--smb

Bob Cain wrote:
> Oh, I failed to ask.  Please, oh please, make it 
> continuously variable under a contour while you're at it.  :-)
> 
> 
> Bob

"David Reid" wrote:
> Well, i think that's what DirectSound allows you to do when you change the
> buffer frequency.  If the sample rate of the sound is 44100 and you multiply
> by 1.08 approx, the buffer will now contain data for 48kHz playback, which
> when played back at 44.1kHz would yield a "faster" sound.  This method was
> acceptable for the company so that's what im trying to achieve but using a
> different sound API (ASIO).  So i have to do the sample rate conversions
> myself.

Yes - and the good thing is you don't need pitch nor frequency
shifting! Just convert the sample rate and there you go...

--smb

Oh, I failed to ask.  Please, oh please, make it 
continuously variable under a contour while you're at it.  :-)


Bob
-- 

"Things should be described as simply as possible, but no 
simpler."

                                              A. Einstein

Stephan M. Bernsee wrote:

> Bob Cain wrote in message news:
> 
>>Good to see you here again Stephan!  Would you care to tell 
>>us where your GPL scaling library stands?
>>
> 
> 
> ...on http://www.clearscale.org...?
> 
>  ;-)
> 
> I'll be working on it over the summer. Right now I have some of my
> students collect some test files and write up a batch script to test
> various methods, which is rather tedious if you do it manually. But I
> already have a working prototype that I'm going to test further and
> I'll update the page as I go.
> 
> Also, I'll most probably be at the Linux Audio developer conference
> held next week in Karlsruhe/ZKM to talk about the latest development
> in this regard.
> 
> The link is http://www.zkm.de/lad (I'm not on the list of speakers
> because I learned about the LAD in February - I'm just there for a
> chat).
> 

Thanks, I can't tell you how much I applaud your willingness 
to take this on.  There is such a dearth of high quality 
public domain technology for this.  Many projects will benefit.


Many Thanks,

Bob
-- 

"Things should be described as simply as possible, but no 
simpler."

                                              A. Einstein

Well, i think that's what DirectSound allows you to do when you change the
buffer frequency.  If the sample rate of the sound is 44100 and you multiply
by 1.08 approx, the buffer will now contain data for 48kHz playback, which
when played back at 44.1kHz would yield a "faster" sound.  This method was
acceptable for the company so that's what im trying to achieve but using a
different sound API (ASIO).  So i have to do the sample rate conversions
myself.

I've started looking at the osalp on sourceforge, it has a class that does
sample rate conversions.  i think that will be good enough for what i want
to do, but we'll see how it goes.

Thanks for all your help.
"Stephan M. Bernsee" <stephan.bernsee@web.de> wrote in message
news:38ab652c.0404210941.59373723@posting.google.com...
> Well, considering the fact that a motor noise is a cyclostationary
> signal rather than a (musically) harmonic signal and since you
> indicate that you wish to increase the "speed" of that cycle, maybe a
> sample rate conversion to transpose it (musically speaking) would be
> better suited?
>
> Pitch shifting will only change the pitch, not the cycle. IOW: it will
> not sound faster, just higher... I'm not sure if that is what you
> want, considering the application...
>
> --smb
>
> "David Reid" wrote in message
> > Yes, thank you stephan.  I meant to write back saying that i want to do
> > pitch shifting.
> >
> > An example of this would be scaling the pitch of an engine sound so that
if
> > the scale factor is > 1 the engine sounds as if its going "faster", and
if
> > it's less than one, it sounds "slower".
> >
> > I've checked out dspdimension, and a few other sites, including
clearscale.
> >
> > David

Well, considering the fact that a motor noise is a cyclostationary
signal rather than a (musically) harmonic signal and since you
indicate that you wish to increase the "speed" of that cycle, maybe a
sample rate conversion to transpose it (musically speaking) would be
better suited?

Pitch shifting will only change the pitch, not the cycle. IOW: it will
not sound faster, just higher... I'm not sure if that is what you
want, considering the application...

--smb

"David Reid" wrote in message 
> Yes, thank you stephan.  I meant to write back saying that i want to do
> pitch shifting.
> 
> An example of this would be scaling the pitch of an engine sound so that if
> the scale factor is > 1 the engine sounds as if its going "faster", and if
> it's less than one, it sounds "slower".
> 
> I've checked out dspdimension, and a few other sites, including clearscale.
> 
> David

Yes, thank you stephan.  I meant to write back saying that i want to do
pitch shifting.

An example of this would be scaling the pitch of an engine sound so that if
the scale factor is > 1 the engine sounds as if its going "faster", and if
it's less than one, it sounds "slower".

I've checked out dspdimension, and a few other sites, including clearscale.

David

"Stephan M. Bernsee" <stephan.bernsee@web.de> wrote in message
news:38ab652c.0404202125.22e6904@posting.google.com...
> David,
>
> first you should be clear about what you actually want to achieve.
>
> A frequency shift shifts all frequencies in a sound by the same amount
> in Hz, distorting it in an "unnatural" way. A pitch shift does not -
> ideally, it will shift the perceived pitch on a note scale to become
> sharp or flat.
>
> Feel free to visit my web site at http://www.dspdimension.com for more
> information.
>
> --smb